1. Field of the Invention
The present invention relates to a method of operating a hearing aid system. The present invention also relates to a hearing aid system adapted to carry out said method.
Within the context of the present disclosure a hearing aid can be understood as a small, battery-powered, microelectronic device designed to be worn behind or in the human ear by a hearing-impaired user. Prior to use, the hearing aid is adjusted by a hearing aid fitter according to a prescription. The prescription is based on a hearing test, resulting in a so-called audiogram, of the performance of the hearing-impaired user's unaided hearing. The prescription is developed to reach a setting where the hearing aid will alleviate a hearing loss by amplifying sound at frequencies in those parts of the audible frequency range where the user suffers a hearing deficit. A hearing aid comprises one or more microphones, a battery, a microelectronic circuit comprising a signal processor adapted to provide amplification in those parts of the audible frequency range where the user suffers a hearing deficit, and an acoustic output transducer. The signal processor is preferably a digital signal processor. The hearing aid is enclosed in a casing suitable for fitting behind or in a human ear.
Within the present context a hearing aid system may comprise a single hearing aid (a so called monaural hearing aid system) or comprise two hearing aids, one for each ear of the hearing aid user (a so called binaural hearing aid system). Furthermore the hearing aid system may comprise an external device, such as a smart phone having software applications adapted to interact with other devices of the hearing aid system. Thus within the present context the term “hearing aid system device” may denote a hearing aid or an external device.
Generally a hearing aid system according to the invention is understood as meaning any system which provides an output signal that can be perceived as an acoustic signal by a user or contributes to providing such an output signal and which has means which are used to compensate for an individual hearing loss of the user or contribute to compensating for the hearing loss of the user. These systems may comprise hearing aids which can be worn on the body or on the head, in particular on or in the ear, and hearing aids that can be fully or partially implanted. However, some devices whose main aim is not to compensate for a hearing loss may nevertheless be considered a hearing aid system, for example consumer electronic devices (televisions, hi-fi systems, mobile phones, MP3 players etc.) provided they have measures for compensating for an individual hearing loss.
Speech enhancement is a fundamental challenge in real-time sound devices such as hearings aids. It is a key reason for hearing impaired people for getting a hearing aid. Traditional speech enhancement or noise suppression techniques consist of splitting the input signals into a number of frequency bands, processing each band according to a selected strategy generally designed to enhance bands carrying speech and to suppress bands carrying noise, and finally combining the bands into a broadband output signal. The width and sharpness of the filters will effectively determine the resolution in time and frequency. Some signal segments consist of narrow frequency components stationary over long periods (e.g., vowels) while other signal segments have a very short duration but span a wide frequency range (e.g., many consonants). If signal components of different types are not processed differently, it is hard to find an appropriate trade-off between resolution in time and resolution in frequency.
In the following, a set-up where noisy speech is processed through a number of fixed filter banks is considered and the inherent limitations of this approach are illustrated. To keep focus on the time- and frequency-resolution of the filter bank, delay constraints are ignored and an ideal Wiener filter is used to process the signal where the noise and speech estimates are obtained from the clean noise and clean speech signals respectively. The analysis window is a Hann window with 50% overlap, and the signal is synthesized using overlap-add. The input signal is speech mixed with speech-shaped noise at different signal-to-noise ratios, and the SNR gain is measured as a function of the length of the analysis window. The results can be seen in FIG. 1. The SNR gain increases as a function of the window length until about 65 miliseconds (ms). For short windows (<10 ms), the sound is heavily affected by musical noise. This is due to statistical variations in the signal estimates, even when the true signals are used. For long windows (>60 ms) the sound has an ‘echo’ effect due to the temporal smearing of the gain envelope. From an energy point of view, a window around 65 ms is optimal since this window length gives a better frequency resolution while not being longer than the long voiced sounds in speech that contain most of the energy in speech. Even though this window length is optimal from an energy point of view, it is usually not a good choice in practice, since it smears transient events like plosives in speech or transition periods.
Therefore a short window is preferred for processing e.g. a ‘t’. The reason why this is not reflected in FIG. 1 is that transients have very little energy compared to the longer voiced sounds even though they are important for speech intelligibility.
Considering the plosive ‘p’ in the beginning of the word ‘puzzle’ a long window will smoothe out the plosive and make the word sound like ‘huzzle’ instead of “puzzle”. This illustrates how long windows can have disastrous results on speech intelligibility because they smear the transients. In practice, a window around 20-30 ms is often chosen as a trade-off between good time resolution and efficient noise suppression arising from a long time window.
Additionally, it is instrumental for the real-time processing carried out in a hearing aid system that the group delay is kept very low to ensure that other people's speech is still perceived as being synchronized with their lip movement and that a user's own speech and sound from the external environment propagating into the ear canal, e.g. through a hearing aid vent, does not get too much out of sync with the sound coming from a hearing aid loudspeaker, whereby a comb-filter effect might result. The choice of filter bank is consequently a fundamental decision for real-time speech enhancement in a hearing aid system as the design is bound to limit some aspects of the performance.
2. The Prior Art
In the paper “Superposition Frames for Adaptive Time-Frequency Analysis and Fast Reconstruction”, by D. Rudoy et. al. in IEEE Transactions on Signal Processing, Vol. 58, 5, May 2010, the tradeoff between time and frequency resolution is addressed by growing a time window by merging the shortest desired windows based on an evaluation of local spectral kurtosis.
In the paper “Improved Reproduction of Stops in Noise Reduction Systems with Adaptive Windows and Nonstationarity Detection” by D. Mauler, R. Martin, in EURASIP Journal on Advances in Signal Processing Volume 2009, Article ID 469480, a real time analysis-synthesis filter bank is developed with a constraint of 10 ms time delay, where a short and a long analysis window is switched depending on the stationarity of the signal.
It is therefore a feature of the present invention to provide a method of operating a hearing system with improved noise suppression.
It is another feature of the present invention to provide a method of adaptive time-frequency analysis using an algorithm having processing power requirements suitable for implementation in a hearing aid system.
It is still another feature of the present invention to provide a method of adaptive time-frequency analysis in a hearing aid system that can be carried out independently for a number of frequency ranges.
It is yet another feature of the present invention to provide a hearing aid system that is adapted to carry out said above mentioned methods.